[Asterisk-Users] Codecs and call failure with Grandstream

[Stephen R. Besch]

I know that this issue has been discussed a lot on this list in regard 
to some of the recent CVS's.  However, it has come up as an issue on an 
older release (CVS Aug 05, 2003) as well. I thought that a heads up was 
in keeping with the philosophy of the list.  Here are the details:

Call from GS via * to remote IAX to PSTN.  Sound stream is established 
from PSTN to GS but no sound from GS to PSTN.

By the way, calls from GS to the PSTN via * worked correctly.  Only the 
IAX bridge failed. It turned out to be a codec problem.  The fix is the 
same as well. Add to sip.conf [general] (or on a phone by phone basis):

disallow=all
allow=alaw
allow=ulaw
You may also need to enable additional codecs.

Stephen R. Besch

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